Script 'mail_helper' called by obssrc Hello community, here is the log from the commit of package qt6-webengine for openSUSE:Factory checked in at 2024-08-13 13:23:00 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ Comparing /work/SRC/openSUSE:Factory/qt6-webengine (Old) and /work/SRC/openSUSE:Factory/.qt6-webengine.new.7232 (New) ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
Package is "qt6-webengine" Tue Aug 13 13:23:00 2024 rev:34 rq:1192339 version:6.7.2 Changes: -------- --- /work/SRC/openSUSE:Factory/qt6-webengine/qt6-webengine.changes 2024-06-24 20:50:23.465940275 +0200 +++ /work/SRC/openSUSE:Factory/.qt6-webengine.new.7232/qt6-webengine.changes 2024-08-13 13:23:08.529991772 +0200 @@ -1,0 +2,6 @@ +Wed Aug 7 12:39:11 UTC 2024 - Christophe Marin <christo...@krop.fr> + +- Add patch to build qtwebengine with ffmpeg 7 (picked from Arch) + * qtwebengine-ffmpeg-7.patch + +------------------------------------------------------------------- New: ---- qtwebengine-ffmpeg-7.patch BETA DEBUG BEGIN: New:- Add patch to build qtwebengine with ffmpeg 7 (picked from Arch) * qtwebengine-ffmpeg-7.patch BETA DEBUG END: ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ Other differences: ------------------ ++++++ qt6-webengine.spec ++++++ --- /var/tmp/diff_new_pack.tEq8Oe/_old 2024-08-13 13:23:11.310107606 +0200 +++ /var/tmp/diff_new_pack.tEq8Oe/_new 2024-08-13 13:23:11.310107606 +0200 @@ -47,6 +47,7 @@ # Patches 0-100 are upstream patches # # Patches 100-200 are openSUSE and/or non-upstream(able) patches # Patch100: rtc-dont-use-h264.patch +Patch101: qtwebengine-ffmpeg-7.patch # # Chromium/blink don't support PowerPC and zSystems and build fails on # 32 bits archs (https://bugreports.qt.io/browse/QTBUG-102143) @@ -345,6 +346,11 @@ %prep %autosetup -p1 -n %{tar_name}-%{real_version}%{tar_suffix} +%if %{pkg_vcmp pkgconfig(libavcodec) <= 5} +# The ffmpeg 7 compatibility patch would break build with older ffmpeg +%patch -P101 -p1 -R +%endif + %build %if %{no_flavor} # Determine the right number of parallel processes based on the available memory ++++++ qtwebengine-ffmpeg-7.patch ++++++ >From 6e554a30893150793c2638e3689cf208ffc8e375 Mon Sep 17 00:00:00 2001 From: Dale Curtis <dalecur...@chromium.org> Date: Sat, 2 Apr 2022 05:13:53 +0000 Subject: [PATCH] Roll src/third_party/ffmpeg/ 574c39cce..32b2d1d526 (1125 commits) https://chromium.googlesource.com/chromium/third_party/ffmpeg.git/+log/574c39cce323..32b2d1d526 Created with: roll-dep src/third_party/ffmpeg Fixed: 1293918 Cq-Include-Trybots: luci.chromium.try:mac_chromium_asan_rel_ng,linux_chromium_asan_rel_ng,linux_chromium_chromeos_asan_rel_ng Change-Id: I41945d0f963e3d1f65940067bac22f63b68e37d2 Reviewed-on: https://chromium-review.googlesource.com/c/chromium/src/+/3565647 Auto-Submit: Dale Curtis <dalecur...@chromium.org> Reviewed-by: Dan Sanders <sande...@chromium.org> Commit-Queue: Dale Curtis <dalecur...@chromium.org> Cr-Commit-Position: refs/heads/main@{#988253} --- .../clear_key_cdm/ffmpeg_cdm_audio_decoder.cc | 29 ++++++++++--------- media/ffmpeg/ffmpeg_common.cc | 11 +++---- media/filters/audio_file_reader.cc | 9 +++--- media/filters/audio_file_reader_unittest.cc | 6 ++-- .../filters/audio_video_metadata_extractor.cc | 11 +++++-- .../filters/ffmpeg_aac_bitstream_converter.cc | 7 +++-- ...ffmpeg_aac_bitstream_converter_unittest.cc | 2 +- media/filters/ffmpeg_audio_decoder.cc | 13 +++++---- 8 files changed, 51 insertions(+), 37 deletions(-) diff --git a/src/3rdparty/chromium/media/cdm/library_cdm/clear_key_cdm/ffmpeg_cdm_audio_decoder.cc b/src/3rdparty/chromium/media/cdm/library_cdm/clear_key_cdm/ffmpeg_cdm_audio_decoder.cc index c535d2b..62ddbc8 100644 --- a/src/3rdparty/chromium/media/cdm/library_cdm/clear_key_cdm/ffmpeg_cdm_audio_decoder.cc +++ b/src/3rdparty/chromium/media/cdm/library_cdm/clear_key_cdm/ffmpeg_cdm_audio_decoder.cc @@ -74,7 +74,7 @@ void CdmAudioDecoderConfigToAVCodecContext( codec_context->sample_fmt = AV_SAMPLE_FMT_NONE; } - codec_context->channels = config.channel_count; + codec_context->ch_layout.nb_channels = config.channel_count; codec_context->sample_rate = config.samples_per_second; if (config.extra_data) { @@ -124,8 +124,8 @@ void CopySamples(cdm::AudioFormat cdm_format, case cdm::kAudioFormatPlanarS16: case cdm::kAudioFormatPlanarF32: { const int decoded_size_per_channel = - decoded_audio_size / av_frame.channels; - for (int i = 0; i < av_frame.channels; ++i) { + decoded_audio_size / av_frame.ch_layout.nb_channels; + for (int i = 0; i < av_frame.ch_layout.nb_channels; ++i) { memcpy(output_buffer, av_frame.extended_data[i], decoded_size_per_channel); output_buffer += decoded_size_per_channel; @@ -185,13 +185,14 @@ bool FFmpegCdmAudioDecoder::Initialize( // Success! decoding_loop_ = std::make_unique<FFmpegDecodingLoop>(codec_context_.get()); samples_per_second_ = config.samples_per_second; - bytes_per_frame_ = codec_context_->channels * config.bits_per_channel / 8; + bytes_per_frame_ = + codec_context_->ch_layout.nb_channels * config.bits_per_channel / 8; output_timestamp_helper_ = std::make_unique<AudioTimestampHelper>(config.samples_per_second); is_initialized_ = true; // Store initial values to guard against midstream configuration changes. - channels_ = codec_context_->channels; + channels_ = codec_context_->ch_layout.nb_channels; av_sample_format_ = codec_context_->sample_fmt; return true; @@ -291,7 +292,8 @@ cdm::Status FFmpegCdmAudioDecoder::DecodeBuffer( for (auto& frame : audio_frames) { int decoded_audio_size = 0; if (frame->sample_rate != samples_per_second_ || - frame->channels != channels_ || frame->format != av_sample_format_) { + frame->ch_layout.nb_channels != channels_ || + frame->format != av_sample_format_) { DLOG(ERROR) << "Unsupported midstream configuration change!" << " Sample Rate: " << frame->sample_rate << " vs " << samples_per_second_ @@ -302,7 +304,7 @@ cdm::Status FFmpegCdmAudioDecoder::DecodeBuffer( } decoded_audio_size = av_samples_get_buffer_size( - nullptr, codec_context_->channels, frame->nb_samples, + nullptr, codec_context_->ch_layout.nb_channels, frame->nb_samples, codec_context_->sample_fmt, 1); if (!decoded_audio_size) continue; @@ -322,7 +324,7 @@ bool FFmpegCdmAudioDecoder::OnNewFrame( std::vector<std::unique_ptr<AVFrame, ScopedPtrAVFreeFrame>>* audio_frames, AVFrame* frame) { *total_size += av_samples_get_buffer_size( - nullptr, codec_context_->channels, frame->nb_samples, + nullptr, codec_context_->ch_layout.nb_channels, frame->nb_samples, codec_context_->sample_fmt, 1); audio_frames->emplace_back(av_frame_clone(frame)); return true; diff --git a/src/3rdparty/chromium/media/ffmpeg/ffmpeg_common.cc b/src/3rdparty/chromium/media/ffmpeg/ffmpeg_common.cc index 2665355..910f9ad 100644 --- a/src/3rdparty/chromium/media/ffmpeg/ffmpeg_common.cc +++ b/src/3rdparty/chromium/media/ffmpeg/ffmpeg_common.cc @@ -336,10 +336,11 @@ bool AVCodecContextToAudioDecoderConfig(const AVCodecContext* codec_context, codec_context->sample_fmt, codec_context->codec_id); ChannelLayout channel_layout = - codec_context->channels > 8 + codec_context->ch_layout.nb_channels > 8 ? CHANNEL_LAYOUT_DISCRETE - : ChannelLayoutToChromeChannelLayout(codec_context->channel_layout, - codec_context->channels); + : ChannelLayoutToChromeChannelLayout( + codec_context->ch_layout.u.mask, + codec_context->ch_layout.nb_channels); switch (codec) { // For AC3/EAC3 we enable only demuxing, but not decoding, so FFmpeg does @@ -391,7 +392,7 @@ bool AVCodecContextToAudioDecoderConfig(const AVCodecContext* codec_context, extra_data, encryption_scheme, seek_preroll, codec_context->delay); if (channel_layout == CHANNEL_LAYOUT_DISCRETE) - config->SetChannelsForDiscrete(codec_context->channels); + config->SetChannelsForDiscrete(codec_context->ch_layout.nb_channels); #if BUILDFLAG(ENABLE_PLATFORM_AC3_EAC3_AUDIO) // These are bitstream formats unknown to ffmpeg, so they don't have @@ -460,7 +461,7 @@ void AudioDecoderConfigToAVCodecContext(const AudioDecoderConfig& config, // TODO(scherkus): should we set |channel_layout|? I'm not sure if FFmpeg uses // said information to decode. - codec_context->channels = config.channels(); + codec_context->ch_layout.nb_channels = config.channels(); codec_context->sample_rate = config.samples_per_second(); if (config.extra_data().empty()) { diff --git a/src/3rdparty/chromium/media/filters/audio_file_reader.cc b/src/3rdparty/chromium/media/filters/audio_file_reader.cc index 777eabc..2b58dd7 100644 --- a/src/3rdparty/chromium/media/filters/audio_file_reader.cc +++ b/src/3rdparty/chromium/media/filters/audio_file_reader.cc @@ -113,14 +113,15 @@ bool AudioFileReader::OpenDecoder() { // Verify the channel layout is supported by Chrome. Acts as a sanity check // against invalid files. See http://crbug.com/171962 - if (ChannelLayoutToChromeChannelLayout(codec_context_->channel_layout, - codec_context_->channels) == + if (ChannelLayoutToChromeChannelLayout( + codec_context_->ch_layout.u.mask, + codec_context_->ch_layout.nb_channels) == CHANNEL_LAYOUT_UNSUPPORTED) { return false; } // Store initial values to guard against midstream configuration changes. - channels_ = codec_context_->channels; + channels_ = codec_context_->ch_layout.nb_channels; audio_codec_ = CodecIDToAudioCodec(codec_context_->codec_id); sample_rate_ = codec_context_->sample_rate; av_sample_format_ = codec_context_->sample_fmt; @@ -223,7 +224,7 @@ bool AudioFileReader::OnNewFrame( if (frames_read < 0) return false; - const int channels = frame->channels; + const int channels = frame->ch_layout.nb_channels; if (frame->sample_rate != sample_rate_ || channels != channels_ || frame->format != av_sample_format_) { DLOG(ERROR) << "Unsupported midstream configuration change!" diff --git a/src/3rdparty/chromium/media/filters/ffmpeg_aac_bitstream_converter.cc b/src/3rdparty/chromium/media/filters/ffmpeg_aac_bitstream_converter.cc index 76b41aa..e26b6cd 100644 --- a/src/3rdparty/chromium/media/filters/ffmpeg_aac_bitstream_converter.cc +++ b/src/3rdparty/chromium/media/filters/ffmpeg_aac_bitstream_converter.cc @@ -195,14 +195,15 @@ bool FFmpegAACBitstreamConverter::ConvertPacket(AVPacket* packet) { if (!header_generated_ || codec_ != stream_codec_parameters_->codec_id || audio_profile_ != stream_codec_parameters_->profile || sample_rate_index_ != sample_rate_index || - channel_configuration_ != stream_codec_parameters_->channels || + channel_configuration_ != + stream_codec_parameters_->ch_layout.nb_channels || frame_length_ != header_plus_packet_size) { header_generated_ = GenerateAdtsHeader(stream_codec_parameters_->codec_id, 0, // layer stream_codec_parameters_->profile, sample_rate_index, 0, // private stream - stream_codec_parameters_->channels, + stream_codec_parameters_->ch_layout.nb_channels, 0, // originality 0, // home 0, // copyrighted_stream @@ -214,7 +215,7 @@ bool FFmpegAACBitstreamConverter::ConvertPacket(AVPacket* packet) { codec_ = stream_codec_parameters_->codec_id; audio_profile_ = stream_codec_parameters_->profile; sample_rate_index_ = sample_rate_index; - channel_configuration_ = stream_codec_parameters_->channels; + channel_configuration_ = stream_codec_parameters_->ch_layout.nb_channels; frame_length_ = header_plus_packet_size; } diff --git a/src/3rdparty/chromium/media/filters/ffmpeg_aac_bitstream_converter_unittest.cc b/src/3rdparty/chromium/media/filters/ffmpeg_aac_bitstream_converter_unittest.cc index 3b46f7f..1897eb0 100644 --- a/src/3rdparty/chromium/media/filters/ffmpeg_aac_bitstream_converter_unittest.cc +++ b/src/3rdparty/chromium/media/filters/ffmpeg_aac_bitstream_converter_unittest.cc @@ -34,7 +34,7 @@ class FFmpegAACBitstreamConverterTest : public testing::Test { memset(&test_parameters_, 0, sizeof(AVCodecParameters)); test_parameters_.codec_id = AV_CODEC_ID_AAC; test_parameters_.profile = FF_PROFILE_AAC_MAIN; - test_parameters_.channels = 2; + test_parameters_.ch_layout.nb_channels = 2; test_parameters_.extradata = extradata_header_; test_parameters_.extradata_size = sizeof(extradata_header_); } diff --git a/src/3rdparty/chromium/media/filters/ffmpeg_audio_decoder.cc b/src/3rdparty/chromium/media/filters/ffmpeg_audio_decoder.cc index bf3ed00..d564ee9 100644 --- a/src/3rdparty/chromium/media/filters/ffmpeg_audio_decoder.cc +++ b/src/3rdparty/chromium/media/filters/ffmpeg_audio_decoder.cc @@ -29,7 +29,7 @@ namespace media { // Return the number of channels from the data in |frame|. static inline int DetermineChannels(AVFrame* frame) { - return frame->channels; + return frame->ch_layout.nb_channels; } // Called by FFmpeg's allocation routine to allocate a buffer. Uses @@ -243,7 +243,7 @@ bool FFmpegAudioDecoder::OnNewFrame(const DecoderBuffer& buffer, // Translate unsupported into discrete layouts for discrete configurations; // ffmpeg does not have a labeled discrete configuration internally. ChannelLayout channel_layout = ChannelLayoutToChromeChannelLayout( - codec_context_->channel_layout, codec_context_->channels); + codec_context_->ch_layout.u.mask, codec_context_->ch_layout.nb_channels); if (channel_layout == CHANNEL_LAYOUT_UNSUPPORTED && config_.channel_layout() == CHANNEL_LAYOUT_DISCRETE) { channel_layout = CHANNEL_LAYOUT_DISCRETE; @@ -360,11 +360,11 @@ bool FFmpegAudioDecoder::ConfigureDecoder(const AudioDecoderConfig& config) { // Success! av_sample_format_ = codec_context_->sample_fmt; - if (codec_context_->channels != config.channels()) { + if (codec_context_->ch_layout.nb_channels != config.channels()) { MEDIA_LOG(ERROR, media_log_) << "Audio configuration specified " << config.channels() << " channels, but FFmpeg thinks the file contains " - << codec_context_->channels << " channels"; + << codec_context_->ch_layout.nb_channels << " channels"; ReleaseFFmpegResources(); state_ = DecoderState::kUninitialized; return false; @@ -415,7 +415,7 @@ int FFmpegAudioDecoder::GetAudioBuffer(struct AVCodecContext* s, if (frame->nb_samples <= 0) return AVERROR(EINVAL); - if (s->channels != channels) { + if (s->ch_layout.nb_channels != channels) { DLOG(ERROR) << "AVCodecContext and AVFrame disagree on channel count."; return AVERROR(EINVAL); } @@ -448,7 +448,8 @@ int FFmpegAudioDecoder::GetAudioBuffer(struct AVCodecContext* s, ChannelLayout channel_layout = config_.channel_layout() == CHANNEL_LAYOUT_DISCRETE ? CHANNEL_LAYOUT_DISCRETE - : ChannelLayoutToChromeChannelLayout(s->channel_layout, s->channels); + : ChannelLayoutToChromeChannelLayout(s->ch_layout.u.mask, + s->ch_layout.nb_channels); if (channel_layout == CHANNEL_LAYOUT_UNSUPPORTED) { DLOG(ERROR) << "Unsupported channel layout."; commit 62274859104bd828373ae406aa9309e610449ac5 Author: Ted Meyer <tmathme...@chromium.org> Date: Fri Mar 22 19:56:55 2024 +0000 Replace deprecated use of AVCodecContext::reordered_opaque We can use the AV_CODEC_FLAG_COPY_OPAQUE flag on the codec context now to trigger timestamp propagation. Bug: 330573128 Change-Id: I6bc57241a35ab5283742aad8d42acb4dc5e85858 Reviewed-on: https://chromium-review.googlesource.com/c/chromium/src/+/5384308 Commit-Queue: Ted (Chromium) Meyer <tmathme...@chromium.org> Reviewed-by: Dan Sanders <sande...@chromium.org> Cr-Commit-Position: refs/heads/main@{#1277051} diff --git a/src/3rdparty/chromium/media/ffmpeg/ffmpeg_common.cc b/src/3rdparty/chromium/media/ffmpeg/ffmpeg_common.cc index 910f9ad..8be165c 100644 --- a/src/3rdparty/chromium/media/ffmpeg/ffmpeg_common.cc +++ b/src/3rdparty/chromium/media/ffmpeg/ffmpeg_common.cc @@ -411,7 +411,9 @@ bool AVCodecContextToAudioDecoderConfig(const AVCodecContext* codec_context, // TODO(dalecurtis): Just use the profile from the codec context if ffmpeg // ever starts supporting xHE-AAC. - if (codec_context->profile == FF_PROFILE_UNKNOWN) { + constexpr uint8_t kXHEAAc = 41; + if (codec_context->profile == FF_PROFILE_UNKNOWN || + codec_context->profile == kXHEAAc) { // Errors aren't fatal here, so just drop any MediaLog messages. NullMediaLog media_log; mp4::AAC aac_parser; diff --git a/src/3rdparty/chromium/media/ffmpeg/ffmpeg_regression_tests.cc b/src/3rdparty/chromium/media/ffmpeg/ffmpeg_regression_tests.cc index 05dcb1c..866f446 100644 --- a/src/3rdparty/chromium/media/ffmpeg/ffmpeg_regression_tests.cc +++ b/src/3rdparty/chromium/media/ffmpeg/ffmpeg_regression_tests.cc @@ -90,16 +90,16 @@ FFMPEG_TEST_CASE(Cr62127, PIPELINE_ERROR_DECODE, PIPELINE_ERROR_DECODE); FFMPEG_TEST_CASE(Cr93620, "security/93620.ogg", PIPELINE_OK, PIPELINE_OK); -FFMPEG_TEST_CASE(Cr100492, - "security/100492.webm", - DECODER_ERROR_NOT_SUPPORTED, - DECODER_ERROR_NOT_SUPPORTED); +FFMPEG_TEST_CASE(Cr100492, "security/100492.webm", PIPELINE_OK, PIPELINE_OK); FFMPEG_TEST_CASE(Cr100543, "security/100543.webm", PIPELINE_OK, PIPELINE_OK); FFMPEG_TEST_CASE(Cr101458, "security/101458.webm", PIPELINE_ERROR_DECODE, PIPELINE_ERROR_DECODE); -FFMPEG_TEST_CASE(Cr108416, "security/108416.webm", PIPELINE_OK, PIPELINE_OK); +FFMPEG_TEST_CASE(Cr108416, + "security/108416.webm", + PIPELINE_ERROR_DECODE, + PIPELINE_ERROR_DECODE); FFMPEG_TEST_CASE(Cr110849, "security/110849.mkv", DEMUXER_ERROR_COULD_NOT_OPEN, @@ -154,7 +154,10 @@ FFMPEG_TEST_CASE(Cr234630b, "security/234630b.mov", DEMUXER_ERROR_NO_SUPPORTED_STREAMS, DEMUXER_ERROR_NO_SUPPORTED_STREAMS); -FFMPEG_TEST_CASE(Cr242786, "security/242786.webm", PIPELINE_OK, PIPELINE_OK); +FFMPEG_TEST_CASE(Cr242786, + "security/242786.webm", + PIPELINE_OK, + PIPELINE_ERROR_DECODE); // Test for out-of-bounds access with slightly corrupt file (detection logic // thinks it's a MONO file, but actually contains STEREO audio). FFMPEG_TEST_CASE(Cr275590, @@ -372,8 +375,8 @@ FFMPEG_TEST_CASE(WEBM_2, DEMUXER_ERROR_NO_SUPPORTED_STREAMS); FFMPEG_TEST_CASE(WEBM_4, "security/out.webm.68798.1929", - DECODER_ERROR_NOT_SUPPORTED, - DECODER_ERROR_NOT_SUPPORTED); + PIPELINE_OK, + PIPELINE_OK); FFMPEG_TEST_CASE(WEBM_5, "frame_size_change.webm", PIPELINE_OK, PIPELINE_OK); // General MKV test cases. diff --git a/src/3rdparty/chromium/media/filters/audio_decoder_unittest.cc b/src/3rdparty/chromium/media/filters/audio_decoder_unittest.cc index a7b2533..ba3c308 100644 --- a/src/3rdparty/chromium/media/filters/audio_decoder_unittest.cc +++ b/src/3rdparty/chromium/media/filters/audio_decoder_unittest.cc @@ -484,7 +484,7 @@ constexpr TestParams kXheAacTestParams[] = { }}, 0, 29400, - CHANNEL_LAYOUT_MONO, + CHANNEL_LAYOUT_UNSUPPORTED, AudioCodecProfile::kXHE_AAC}, #endif {AudioCodec::kAAC, diff --git a/src/3rdparty/chromium/media/filters/audio_file_reader.cc b/src/3rdparty/chromium/media/filters/audio_file_reader.cc index 2b58dd7..9d37f32 100644 --- a/src/3rdparty/chromium/media/filters/audio_file_reader.cc +++ b/src/3rdparty/chromium/media/filters/audio_file_reader.cc @@ -243,18 +243,10 @@ bool AudioFileReader::OnNewFrame( // silence from being output. In the case where we are also discarding some // portion of the packet (as indicated by a negative pts), we further want to // adjust the duration downward by however much exists before zero. -#if BUILDFLAG(USE_SYSTEM_FFMPEG) - if (audio_codec_ == AudioCodec::kAAC && frame->pkt_duration) { -#else if (audio_codec_ == AudioCodec::kAAC && frame->duration) { -#endif // BUILDFLAG(USE_SYSTEM_FFMPEG) const base::TimeDelta pkt_duration = ConvertFromTimeBase( glue_->format_context()->streams[stream_index_]->time_base, -#if BUILDFLAG(USE_SYSTEM_FFMPEG) - frame->pkt_duration + std::min(static_cast<int64_t>(0), frame->pts)); -#else frame->duration + std::min(static_cast<int64_t>(0), frame->pts)); -#endif // BUILDFLAG(USE_SYSTEM_FFMPEG) const base::TimeDelta frame_duration = base::Seconds(frames_read / static_cast<double>(sample_rate_)); diff --git a/src/3rdparty/chromium/media/filters/audio_file_reader_unittest.cc b/src/3rdparty/chromium/media/filters/audio_file_reader_unittest.cc index a1c633d..5784fe1 100644 --- a/src/3rdparty/chromium/media/filters/audio_file_reader_unittest.cc +++ b/src/3rdparty/chromium/media/filters/audio_file_reader_unittest.cc @@ -61,15 +61,14 @@ class AudioFileReaderTest : public testing::Test { // Verify packets are consistent across demuxer runs. Reads the first few // packets and then seeks back to the start timestamp and verifies that the // hashes match on the packets just read. - void VerifyPackets() { - const int kReads = 3; + void VerifyPackets(int packet_reads) { const int kTestPasses = 2; AVPacket packet; base::TimeDelta start_timestamp; std::vector<std::string> packet_md5_hashes_; for (int i = 0; i < kTestPasses; ++i) { - for (int j = 0; j < kReads; ++j) { + for (int j = 0; j < packet_reads; ++j) { ASSERT_TRUE(reader_->ReadPacketForTesting(&packet)); // On the first pass save the MD5 hash of each packet, on subsequent @@ -98,7 +97,8 @@ class AudioFileReaderTest : public testing::Test { int sample_rate, base::TimeDelta duration, int frames, - int expected_frames) { + int expected_frames, + int packet_reads = 3) { Initialize(fn); ASSERT_TRUE(reader_->Open()); EXPECT_EQ(channels, reader_->channels()); @@ -112,7 +112,7 @@ class AudioFileReaderTest : public testing::Test { EXPECT_EQ(reader_->HasKnownDuration(), false); } if (!packet_verification_disabled_) - ASSERT_NO_FATAL_FAILURE(VerifyPackets()); + ASSERT_NO_FATAL_FAILURE(VerifyPackets(packet_reads)); ReadAndVerify(hash, expected_frames); } @@ -219,7 +219,7 @@ TEST_F(AudioFileReaderTest, AAC_ADTS) { } TEST_F(AudioFileReaderTest, MidStreamConfigChangesFail) { - RunTestFailingDecode("midstream_config_change.mp3", 42624); + RunTestFailingDecode("midstream_config_change.mp3", 0); } #endif @@ -229,7 +229,7 @@ TEST_F(AudioFileReaderTest, VorbisInvalidChannelLayout) { TEST_F(AudioFileReaderTest, WaveValidFourChannelLayout) { RunTest("4ch.wav", "131.71,38.02,130.31,44.89,135.98,42.52,", 4, 44100, - base::Microseconds(100001), 4411, 4410); + base::Microseconds(100001), 4411, 4410, /*packet_reads=*/2); } TEST_F(AudioFileReaderTest, ReadPartialMP3) { diff --git a/src/3rdparty/chromium/media/filters/ffmpeg_video_decoder.cc b/src/3rdparty/chromium/media/filters/ffmpeg_video_decoder.cc index e62b2af..ab39796 100644 --- a/src/3rdparty/chromium/media/filters/ffmpeg_video_decoder.cc +++ b/src/3rdparty/chromium/media/filters/ffmpeg_video_decoder.cc @@ -125,7 +125,7 @@ bool FFmpegVideoDecoder::IsCodecSupported(VideoCodec codec) { } FFmpegVideoDecoder::FFmpegVideoDecoder(MediaLog* media_log) - : media_log_(media_log) { + : media_log_(media_log), timestamp_map_(128) { DVLOG(1) << __func__; DETACH_FROM_SEQUENCE(sequence_checker_); } @@ -204,10 +204,6 @@ int FFmpegVideoDecoder::GetVideoBuffer(struct AVCodecContext* codec_context, frame->linesize[plane] = layout->planes()[plane].stride; } - // This seems unsafe, given threaded decoding. However, `reordered_opaque` is - // also going away upstream, so we need a whole new mechanism either way. - frame->reordered_opaque = codec_context->reordered_opaque; - // This will be freed by `ReleaseVideoBufferImpl`. auto* opaque = new OpaqueData(fb_priv, frame_pool_, data, allocation_size, std::move(*layout)); @@ -354,8 +350,10 @@ bool FFmpegVideoDecoder::FFmpegDecode(const DecoderBuffer& buffer) { DCHECK(packet->data); DCHECK_GT(packet->size, 0); - // Let FFmpeg handle presentation timestamp reordering. - codec_context_->reordered_opaque = buffer.timestamp().InMicroseconds(); + const int64_t timestamp = buffer.timestamp().InMicroseconds(); + const TimestampId timestamp_id = timestamp_id_generator_.GenerateNextId(); + timestamp_map_.Put(std::make_pair(timestamp_id, timestamp)); + packet->opaque = reinterpret_cast<void*>(timestamp_id.GetUnsafeValue()); } FFmpegDecodingLoop::DecodeStatus decode_status = decoding_loop_->DecodePacket( packet, base::BindRepeating(&FFmpegVideoDecoder::OnNewFrame, @@ -414,7 +412,12 @@ bool FFmpegVideoDecoder::OnNewFrame(AVFrame* frame) { } gfx::Size natural_size = aspect_ratio.GetNaturalSize(visible_rect); - const auto pts = base::Microseconds(frame->reordered_opaque); + const auto ts_id = TimestampId(reinterpret_cast<size_t>(frame->opaque)); + const auto ts_lookup = timestamp_map_.Get(ts_id); + if (ts_lookup == timestamp_map_.end()) { + return false; + } + const auto pts = base::Microseconds(std::get<1>(*ts_lookup)); auto video_frame = VideoFrame::WrapExternalDataWithLayout( opaque->layout, visible_rect, natural_size, opaque->data, opaque->size, pts); @@ -489,8 +492,10 @@ bool FFmpegVideoDecoder::ConfigureDecoder(const VideoDecoderConfig& config, codec_context_->thread_count = GetFFmpegVideoDecoderThreadCount(config); codec_context_->thread_type = FF_THREAD_SLICE | (low_delay ? 0 : FF_THREAD_FRAME); + codec_context_->opaque = this; codec_context_->get_buffer2 = GetVideoBufferImpl; + codec_context_->flags |= AV_CODEC_FLAG_COPY_OPAQUE; if (decode_nalus_) codec_context_->flags2 |= AV_CODEC_FLAG2_CHUNKS; diff --git a/src/3rdparty/chromium/media/filters/ffmpeg_video_decoder.h b/src/3rdparty/chromium/media/filters/ffmpeg_video_decoder.h index 60cb9d5..4fa8628 100644 --- a/src/3rdparty/chromium/media/filters/ffmpeg_video_decoder.h +++ b/src/3rdparty/chromium/media/filters/ffmpeg_video_decoder.h @@ -7,10 +7,12 @@ #include <memory> +#include "base/containers/lru_cache.h" #include "base/functional/callback.h" #include "base/memory/raw_ptr.h" #include "base/memory/scoped_refptr.h" #include "base/sequence_checker.h" +#include "base/types/id_type.h" #include "media/base/supported_video_decoder_config.h" #include "media/base/video_decoder.h" #include "media/base/video_decoder_config.h" @@ -87,6 +89,20 @@ class MEDIA_EXPORT FFmpegVideoDecoder : public VideoDecoder { // FFmpeg structures owned by this object. std::unique_ptr<AVCodecContext, ScopedPtrAVFreeContext> codec_context_; + // The gist here is that timestamps need to be 64 bits to store microsecond + // precision. A 32 bit integer would overflow at ~35 minutes at this level of + // precision. We can't cast the timestamp to the void ptr object used by the + // opaque field in ffmpeg then, because it would lose data on a 32 bit build. + // However, we don't actually have 2^31 timestamped frames in a single + // playback, so it's fine to use the 32 bit value as a key in a map which + // contains the actual timestamps. Additionally, we've in the past set 128 + // outstanding frames for re-ordering as a limit for cross-thread decoding + // tasks, so we'll do that here too with the LRU cache. + using TimestampId = base::IdType<int64_t, size_t, 0>; + + TimestampId::Generator timestamp_id_generator_; + base::LRUCache<TimestampId, int64_t> timestamp_map_; + VideoDecoderConfig config_; scoped_refptr<FrameBufferPool> frame_pool_; diff --git a/src/3rdparty/chromium/third_party/webrtc/modules/video_coding/codecs/h264/h264_decoder_impl.cc b/src/3rdparty/chromium/third_party/webrtc/modules/video_coding/codecs/h264/h264_decoder_impl.cc index f67718c..fe42aef 100644 --- a/src/3rdparty/chromium/third_party/webrtc/modules/video_coding/codecs/h264/h264_decoder_impl.cc +++ b/src/3rdparty/chromium/third_party/webrtc/modules/video_coding/codecs/h264/h264_decoder_impl.cc @@ -229,7 +229,6 @@ int H264DecoderImpl::AVGetBuffer2(AVCodecContext* context, int total_size = y_size + 2 * uv_size; av_frame->format = context->pix_fmt; - av_frame->reordered_opaque = context->reordered_opaque; // Create a VideoFrame object, to keep a reference to the buffer. // TODO(nisse): The VideoFrame's timestamp and rotation info is not used. @@ -377,8 +376,6 @@ int32_t H264DecoderImpl::Decode(const EncodedImage& input_image, return WEBRTC_VIDEO_CODEC_ERROR; } packet->size = static_cast<int>(input_image.size()); - int64_t frame_timestamp_us = input_image.ntp_time_ms_ * 1000; // ms -> μs - av_context_->reordered_opaque = frame_timestamp_us; int result = avcodec_send_packet(av_context_.get(), packet.get()); @@ -395,10 +392,6 @@ int32_t H264DecoderImpl::Decode(const EncodedImage& input_image, return WEBRTC_VIDEO_CODEC_ERROR; } - // We don't expect reordering. Decoded frame timestamp should match - // the input one. - RTC_DCHECK_EQ(av_frame_->reordered_opaque, frame_timestamp_us); - // TODO(sakal): Maybe it is possible to get QP directly from FFmpeg. h264_bitstream_parser_.ParseBitstream(input_image); absl::optional<int> qp = h264_bitstream_parser_.GetLastSliceQp();
